Problem with site-to-site extension dialing on UC520 with IOS 12.4(20)T3
UPDATE: I finally got this problem fixed. It ended up being a problem in my router configuration. Here is the configuration that solved my problem:
interface Tunnel0 ip unnumbered BVI1(must be set to BVI1 or Vlan1 and not FastEthernet0/0) tunnel source FastEthernet0/0 tunnel destination<IP address of other UC520 unit> tunnel mode ipsec ipv4 tunnel protection ipsec profile<IPsec profile for site-to-site VPN>
We have a customer who is having problems transferring or conferencing calls between sites. I have configured site-to-site extension dialing between the two sites, but there is a problem transferring calls or conferencing between the two sites.
Both sites have 8-user UC520 units with IOS 12.4(20)T3. Extensions at site 1 are in the range between 200 and 299. Extensions at site 2 are in the range between 300 and 399. I would like to be able to use dial peers to do 3-digit extension dialing from site 1 to site 2, and to do 3-digit extension dialing from site 2 to site 1. How can I accomplish this and solve the problem with transferring calls or conferencing?
Whenever a person is transferred from one site to the other site with site-to-site extension dialing, they are experiencing one-way audio. Is there a way to fix the one-way audio problem?
We also have another customer, who is going to be converted from Linksys One to SBCS, who needs to have this feature configured, and we really need to have this working on the UC520.
In the dial peers below and in the sample configs that I have attached, I have not used the real SIP proxy parameters, IP addresses, or names.
In the examples, Site 1 has the following IP settings:
WAN IP: 184.108.40.206
Data VLAN: 192.168.10.0/24
Voice VLAN: 10.1.1.0/24
CUE Subnet: 10.1.10.1/30
In the examples, Site 2 has the following IP settings:
WAN IP: 220.127.116.11
Data VLAN: 192.168.11.0/24
Voice VLAN: 10.1.2.0/24
CUE Subnet: 10.1.11.1/30
Here is the dial peer for inbound calls:
dial-peer voice 1000 voip description ** Incoming call from SIP trunk ** voice-class codec 1 voice-class sip dtmf-relay force rtp-nte session protocol sipv2 session target sip-server incoming called-number .% dtmf-relay rtp-nte ip qos dscp cs5 media ip qos dscp cs4 signaling no vad
Here is the dial peer for extension dialing from Site 1 to Site 2: dial-peer voice 5000 voip destination-pattern 3.. voice-class codec 1 voice-class sip dtmf-relay force rtp-nte session protocol sipv2 session target ipv4:18.104.22.168 dtmf-relay rtp-nte ip qos dscp cs5 media ip qos dscp cs4 signaling no vad
Here is the dial peer for extension dialing from Site 2 to Site 1:
! dial-peer voice 5000 voip destination-pattern 2.. voice-class codec 1 voice-class sip dtmf-relay force rtp-nte session protocol sipv2 session target ipv4:22.214.171.124 dtmf-relay rtp-nte ip qos dscp cs5 media ip qos dscp cs4 signaling no vad
What settings do I need to add to the UC520 configurations to fix the site-to-site call transfer and conferencing problems?
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